US6496794B1 - Method and apparatus for seamless multi-rate speech coding - Google Patents

Abstract

A communications system (100) includes a multi-rate source coder (MRSC) (102), a variable size/rate buffer (VSRB) (112), a speech buffer (104), and a buffer control block (106). The variable size/rate buffer (112) includes a source coder bit buffer (SCBB) (114) and an adaptive transmit frame buffer (116). The source coder bit buffer (114) receives speech frames coded at different rates from the multi-rate source coder (102), and deposits an integer or non-integer number of frames in the adaptive transmit frame buffer (ATFB) (116). A receiver includes a seamless rate transition module (SRTM) (308) and an variable buffer (310). The seamless rate transition module (308) correlates speech data previously coded at different rates, and it then truncates or alternatively appends, concatenates, and warps the speech data to remove any annoying artifacts at the rate change boundary.

Description

GOVERNMENT LICENSE RIGHTS

The U.S. Government has a paid-up license in this invention and the right in limited circumstances to require the patent owner to license others on reasonable terms as provided for by the terms of Contract No. DAAL01-96-2-0002 awarded by the U.S. Army.

FIELD OF THE INVENTION

The present invention relates generally to communications systems and, in particular, to multi-rate speech coding systems.

BACKGROUND OF THE INVENTION

Good quality voice services are in high demand, due in part to the emergence of global communication capabilities, such as those provided by cellular systems, satellite systems, landline systems, wireless systems, and combinations thereof. Digital speech coders typically used in these types of systems often operate at fixed rates that utilize a given amount of channel bandwidth. When enough channel bandwidth is available, fixed rate speech coders provide good quality voice services.

The transmission channel medium, however, is often capacity limited or causes excessively high bit error rates. When channel capacity changes, fixed rate coders are often unable to provide synthesized speech at a fixed delay, and they cannot dedicate additional forward error correction bits for protection against the noisy channel. In wireless applications, the channel capacity can change dramatically, and it thus imposes a variable limit on the maximum bit rate that can be passed through the channel.

Variable rate speech coders can reduce the coding rate when channel capacity diminishes, but the quality of speech can suffer. The quality suffers in part because of “artifacts” in the synthesized waveform at the boundary between coding rates. For example, when the variable rate speech coder changes from one coding rate to another, a user may experience a “pop” sound or a silent period due to a discontinuity in the synthesized speech waveform.

A significant need therefore exists for an improved method and apparatus for providing speech coding on a variable bandwidth channel.

BRIEF DESCRIPTION OF THE DRAWINGS

The invention is pointed out with particularity in the appended claims. However, a more complete understanding of the present invention may be derived by referring to the detailed description and claims when considered in connection with the figures, wherein like reference numbers refer to similar items throughout the figures, and:

FIG. 1 shows a communications system in accordance with a preferred embodiment of the present invention;

FIG. 2 is a flowchart of a method for operating a variable size/rate buffer in accordance with a preferred embodiment of the present invention;

FIG. 3 shows a portion of a receiver in accordance with a preferred embodiment of the present invention;

FIG. 4 shows a variable buffer in accordance with a preferred embodiment of the present invention;

FIG. 5 shows a warping factor function in accordance with a preferred embodiment of the present invention;

FIG. 6 is a flowchart of a method for operating a seamless rate transition module in accordance with a preferred embodiment of the present invention;

FIG. 7 is a graph of speech distortion waveforms; and

FIG. 8 is a graph of delay probabilities.

DETAILED DESCRIPTION OF THE DRAWINGS

In the following detailed description, reference is made to the accompanying drawings that show, by way of illustration, specific embodiments in which the invention may be practiced. It is to be understood that other embodiments may be utilized and structural changes may be made without departing from the scope of the present invention.

The method and apparatus of the present invention provide a multi-rate speech coding mechanism that can seamlessly change coding rates “on the fly.” Rate change requests can be generated by the communication system, or they can be generated in response to changing channel characteristics. For example, when fading occurs in the channel, the coding rate can be reduced to allow for either additional forward error correction, or a reduction in the modem symbol rate. The multi-rate speech coding mechanism is switchable between different rates that can be requested at any time, and it produces smooth transitions at the switch locations without producing annoying artifacts in the speech.

Turning now to the drawings in which like reference characters indicate corresponding elements throughout the several views, attention is first directed to FIG.1. FIG. 1 shows a communications system in accordance with a preferred embodiment of the present invention.Communications system100 comprises a transmitter portion,variable capacity channel120 and a receiver portion. The transmitter portion ofcommunications system100 includes multi-rate source coder (MRSC)102, speech buffer104,buffer control block106, and variable size/rate buffer (VSRB)112. The receiver portion includes multi-rate source (de)coder MRSC130, speech buffer132,buffer control block134, andVSRB122.

MRSC102 produces “frames” of coded speech. Speech present onspeech input node108 is divided into discrete segments, each segment being one “frame size” in time duration. Frames of coded speech produced byMRSC102 hold digital information (bits), and the number of bits per frame is a function of the frame size and coding rate. When the coding rate changes, the frame size can change, and the number of bits per frame can change.

The transmitter portion ofcommunications system100 outputs “blocks” of coded speech tovariable capacity channel120. A block can be an integer number of frames, or a non-integer number of frames. The receiver portion ofcommunications system100 receives blocks fromvariable capacity channel120, andMRSC130 processes frames of coded speech within the blocks.

MRSC102 can be any type of multi-rate coder, such as a multi-mode code excited linear predictive (CELP) coder, or a multi-rate multiband excitation (MBE) speech coder. In a preferred embodiment,MRSC102 is a multi-rate sinusoidal transform coder (MRSTC). TheMRSC102 can also comprise multiple types of speech coders, such as CELP at 9.6 kbps, MBE at 4.8 kbps, sinusoidal transform coder at 2.4 kbps, or the like. The MRSTC is preferably a modular MRSTC that is optimized for each coding rate. Any number of different coding rates can be used; however, in a preferred embodiment, four bit rates are used. They are 9.6 kilobits/second (kb/s), 4.8 kb/s, 2.4 kb/s, and 1.2 kb/s. One advantage of utilizing a modular MRSTC forMRSC102 is to increase speech quality at each bit rate without a corresponding increase in algorithmic complexity. The interface provides the ability to switch between any of the four rates, at any time, without producing annoying artifacts at the switch locations. The modular MRSTC produces a graceful degradation in speech quality as the rate decreases.

The sinusoidal transform analysis and synthesis blocks can be used at any of the desired rates with slight differences in the algorithm at each rate. One difference is in the parameters used to perform the signal processing; for example, linear predictive (LP) analysis order changes with the rate. The coding/decoding blocks are rate specific, because unique quantization codebooks are used at each rate to produce good speech quality at the lower encoding rates. Table 1 provides a summary of exemplary MRSTC algorithmic details at each of the four rates.

TABLE 1
Multi-rate voice coder parameters.

	Bit Rate	Frame Size	Bits/	LFC
	(kb/s)	(msec)	Frame	Order
	1.2	40	48	10
	2.4	30	72	14
	4.8	30	144	16
	9.6	25	240	16

MRSC102 changes the coding rate in response to a rate change request on ratechange request node110. MRSC102 can be used in different modes, including a network-controlled mode and a channel-controlled mode. In network-controlled mode,MRSC102 switches between any rate at the request of a network rate control signal on ratechange request node110. In channel-controlled mode, similar switching is provided, but in response to changing channel conditions, as determined by the communication system. For example, in a wireless communication system, long-term fading can cause very low received signal-to-noise ratios, resulting in excessively high bit error rates over long time duration. To reduce the speech distortion due to uncorrected bit errors, the system requests a reduction in the speech coding bit rate. In addition, the system can request an increase in forward error correction bit rate or a reduction in the modem symbol rate, or both.

Buffer control block106 processes the rate request on ratechange request node110, and it passes the appropriate control parameters toMRSC102, speech buffer104, andVSRB112. In some embodiments, there exists a time delay between receipt of a rate change request bybuffer control block106 and whenMRSC102 changes the coding rate. This is due in part to the time left to finish coding the current speech frame. In some embodiments, this delay can be reduced or eliminated by ignoring the current speech frame, and backing up the appropriate amount in the speech buffer to begin coding at the new rate. In some embodiments, the rate change request is sent to the receiver via a low-bandwidth side information channel. In other embodiments, a separate field carries rate information as part of the transmitted frame structure.

Speech buffer104 at the transmitter operates slightly differently than speech buffer132 at the receiver. At the transmitter, speech buffer104 stores past samples of digitized speech. This helps to smooth the resulting synthesized waveforms during rate changes and also helps to reduce the time delay between a rate change request and when the rate change occurs. Speech buffer132 at the receiver helps to remove jitter due to variance in data delivery rates.

In some embodiments, speech buffer104 is not present. In these embodiments, speech data is not buffered prior to coding. WhenMRSC102 receives a rate change request, a time delay may exist between the time the rate change request is received and the time at which the rate change takes place.

VSRB112 includes source coder bit buffer (SCBB)114 and adaptive transmit frame buffer (ATFB)116. One function ofATFB116 is to allow variable block sizes of bits to be transmitted. This aids in reducing end-to-end delay of digital voice data transmitted over the Internet, and it supports adaptive-rate modulation for transmission of digital voice data over wireless channels.

SCBB114 receives coded frames fromMRSC102 and transfers them to ATFB116 for transmission.SCBB114 can receive consecutive frames coded at different rates. For example,SCBB114 is shown in FIG. 1 as having three frames therein. Two frames (frames N and N+1) are coded at an “old” rate (rate1), and one frame (frame1) is coded at a “new” rate (rate2).

In a preferred embodiment, the block size that is transmitted is set to be directly proportional to the rate ofMRSC102. The ATFB frame delivery rate is then proportional to the time taken to fillSCBB114 with an integer number of speech frames, I_fat the current source coding rate. The frame delivery rate is not restricted to a fixed value, in part because correct timing coordination is selected betweenATFB116 andSCBB114. When I_fis reached,buffer control block106 sends out a control signal indicating it is time to transfer I_fframes to ATFB116 and to output the block of bits fromVSRB112.

The data flow at the receiver is in general the reverse of that at the transmitter.VSRB122 receives blocks of bits fromvariable capacity channel120. The blocks are received into adaptive receive frame buffer (ARFB)126. When an appropriate number of blocks has been received, the blocks are transferred to source decoder bit buffer (SDBB)124. Frames of coded speech are sent fromSDBB124 toMRSC130 for decoding. Frames of decoded speech are sent fromMRSC130 to speech buffer132, from which speech data is output onspeech output node136. Speech buffer132 can include a seamless rate transition module (SRTM) and variable buffer as explained more fully below with reference to FIG.3.

One advantage of usingVSRB112 can be seen by showing the end-to-end delay compared to a fixed size/rate buffering (FSRB) approach, which is typically used for fixed-symbol rate wireless systems. For the purposes of this comparison, an assumption is made that for the FSRB the transmit frame buffer holds a fixed number of bits, the frame transmit rate is fixed, but storage variance is allowed in the number and size of source coder frames (same as the VSRB). Also for purposes of this comparison, an assumption is made that the output bit rate of the VSRB is equal to the source coder bit rate. The total delay, t_dv, for the VSRB can be written as

t_dv=t_sw+t_fsr=(t_fsc−t_req)+t_fsr(msec),  (1)

where t_fscis the vocoder frame size at the current rate, t_fsris the vocoder frame size at the new rate, t_reqis the time of the “rate change request,” relative to the end of the current frame boundary, and t_swis the time difference between t_fscand t_req. We assume that t_reqoccurs such that it is uniformly distributed within a frame of length equal to t_fsc. The total delay, t_df, for the FSRB is

t_df=t_sw+t_buf(msec),  (2)

where t_swis as defined above with a multiplication factor of B_t/B_vo, which is a modifier that represents the number of vocoder frames taken to fill transmit frame buffer, B_vois the vocoder frame size at the current rate (bits per frame), and t_bufis the time required to fill the transmit frame buffer. t_bufcan be expressed as

t_buf=(B_t/B_v)t_fsr(msec),  (3)

where B_tis the transmit frame buffer size (bits per frame), B_vis the vocoder frame size at the new rate (also in bits per frame). In the embodiment corresponding to the above equations, the transmit frame buffer holds an integer number of vocoder speech frames. In other embodiments, the transmit frame buffer holds a non-integer number of vocoder speech frames.

FIG. 2 is a flowchart of a method for operating a variable size/rate buffer in accordance with a preferred embodiment of the present invention.Method200 begins inblock202 where the VSRB is initialized on system power up. Initialized parameters include: previous speech encoding bit rate request value (R₀); new speech encoding bit rate request value (R); the size (number) of bits in the ATFB (B_ATFS); the previous B_ATFSor (B_ATFS0); the number of speech frames utilized to create the first frame of speech at the new rate (HIST); and bits previously stored in the bit buffer (B₀).

The encoding rate is set to R, and then inblock204, a determination is made whether R is equal to the old encoding rate. If R is equal to the old rate, then there is no change in rate, and coding continues at the same rate inblock206. If R is not equal to the old rate, then there is a change in rate, andmethod200 transitions fromblock204 to block208 where the current frame is finished coding at the old rate. Inblock210, a buffer control flag (F_R), indicating the rate change location in the bit buffer, is sent to the buffer controller, such as buffer control block106 (FIG.1). F_Rcan mark the location of the end of the last frame coded at the old rate, or it can mark the location of the beginning of the first frame coded at the new rate. Inblock212, voice coding at rate R begins HIST speech frames back in time by utilizing data in the speech buffer and sending the voicing probability (VP) to a voicing memory.

From either block206 or block212,method200 transitions to block214 where the quantized bits are sent to the bit buffer, and B_ATFSand F_Rare read from the buffer controller. Inblock216, a determination is made whether there is a change in rate. If there is no change in rate, thenmethod200 transitions to block220 where a determination is made whether the number of bits in the bit buffer is sufficient to be transmitted at the old encoding rate, e.g., is B(F_R)≧B_ATFS0? If the number of bits in the bit buffer is not sufficient, thenmethod200 transitions to block226 where it remains until sufficient bits exist in the buffer. Otherwise,method200 transitions to block228 where B_ATFSbits are transferred from the bit buffer to the ATFB.

If, inblock216, it is determined that a change in rate has occurred,method200 transitions to block218 where a determination is made whether the current region of speech is unvoiced or silent, e.g., is VP (voicing probability)<1/4? Known mechanisms for determining if speech in the current frame is voiced or unvoiced can be utilized. If true (unvoiced or silent region), it is determined whether the number of bits in the bit buffer is sufficient to be transmitted, e.g., is (B(F_R)≧B_ATFS0) at the current rate inblock220. If true, B_ATFSbits are transferred to the ATFB inblock228, and the bits are transmitted, R is set to R₀, and B_ATFS0is set to B_ATFSinblock230. If false, encoding is continued inblock226 until there are enough bits in the bit buffer for transfer, e.g., until B(F_R)≧B_ATFS. Method200 then continues inblock228 as before.

If, inblock218, it is determined that the current region of speech is a voiced region (i.e. VP>1/4), a smooth transition without artifacts is provided in the transitioned speech region.Method200 proceeds by making the determination of how many bits are in the bit buffer, e.g., is B(F_R−B_MAX)≧B_ATFS0at the old rate inblock222 or at the new rate inblock224. If true, B_ATFS0bits are transferred to the ATFB in block228 (some leftover from the previous rate), the bits are transmitted, R is set to R₀, and B_ATFS0is set to B_ATFSinblock230. If false, encoding is continued inblock226 until there are enough bits in the bit buffer for transfer, e.g., until B(F_R)>B_ATFS. Method200 continues when B_ATFSbits are transferred to the ATFB inblock228, and the bits are transmitted, R is set to R₀, and B_ATFS0is set to B_ATFSinblock230.Method200 continues by transitioning to block204, unless the communication is terminated or there is no more speech to encode.

Method200 represents a particular embodiment where voice coding for the current frame is finished at R₀if the rate change request occurs prior to the end of the current frame. In other embodiments, the last frame at the old rate is dropped, and past digitized speech samples are used. The past digitized samples are coded at the new rate, and the transitions are sewn together using one frame of past speech coded at the new rate.

FIG. 3 shows a portion of a receiver in accordance with a preferred embodiment of the present invention.Receiver300 includesVSRB304,MRSC306, seamless rate transition module (SRTM)308, andvariable buffer310.VSRB304 can be a VSRB such as VSRB122 (FIG.1).MRSC306 is a multi-rate (de)coder, such as MRSC130 (FIG.1).SRTM308 andvariable buffer310 work together to provide a “seamless” transition from speech data that was coded at one rate to speech data that was coded at another rate. Speech data received atSRTM308 is synthesized byMRSC306. When synthesized speech data that was previously coded at one rate is concatenated with synthesized speech data that was previously coded at another rate, a discontinuity and annoying artifacts in the resultant speech waveform can result.SRTM308 andvariable buffer310 operate together to remove discontinuities and annoying artifacts.

FIG. 4 shows a variable buffer in accordance with a preferred embodiment of the present invention.Variable buffer310 can hold a variable number of speech samples. The current number of samples invariable buffer310 is denoted by the value N_DA. N_DA406 can vary betweenN_DAMIN408 andN_DAMAX404 while supplying a steady stream of speech data onnode312. It is desirable to supply a steady stream of speech data in part because ifvariable buffer310 is allowed to underflow, the speech data onnode312 will stop and if the speech data is being sent to a digital-to-analog (D/A) converter, a discontinuity will result. It is also desirable to not letvariable buffer310 overflow, in part because speech data will be lost.

Variable buffer310 receives speech data fromSRTM308 onnode402. WhenN_DA406 is approaching N_DAMIN, it may be desirable to expand, or “warp,” the speech data onnode402 such that the speech data will take up more room invariable buffer310. When warping speech data, a “warping factor” can be found that determines the amount of warping applied to speech data. In addition, ifN_DA406 is approaching N_DAMAX, it may be desirable to compress, or “warp, ” the speech data onnode402 such that the speech data will take up less room invariable buffer310.

FIG. 5 shows a warping factor function in accordance with a preferred embodiment of the present invention. Warping factor (W_f) can take on values ranging from zero to the value of the expansion factor (XF). When N_DAis equal to N_DAMAX, W_fis equal to zero, signifying no expansion. When N_DAis equal to N_DAMIN, W_fis equal to XF, signifying full expansion. FIG. 5 shows a linear warping factor for one embodiment of the invention. In other embodiments, the warping factor is a non-linear function of the number of speech samples. In these embodiments, the warping factor exhibits a curved shape rather than a straight line as shown in FIG.5. In the embodiment shown in FIG. 5, the warping factor can be found as:

W_f=XF*(1−[N_DA−N_DAMIN]/[N_DAMAX−N_DAMIN])  (4)

FIG. 6 is a flowchart of a method for operating a seamless rate transition module (SRTM) in accordance with a preferred embodiment of the present invention.Method600 is invoked when frames received by the SRTM are coded at different rates.Method600 serves to seamlessly transition between two different rates of synthesized speech that may or may not be pitch synchronous, or that exhibit good likeness properties when audibly heard in succession. When merged bymethod600, the speech does not exhibit annoying artifacts. Inblock615, the speech pitch is determined, where the speech pitch has a period (P) associated therewith. Because the old rate is likely to have the most stable speech parameters (due to it having been run for some period of time longer than the coder at the new rate), the speech pitch is determined at the old rate. In a preferred embodiment, the speech pitch is determined using an absolute magnitude difference function (AMDF) on the last frame of speech at the old rate; however, any appropriate pitch determination method can be utilized.

Inblock620, a short-term speech segment is then assigned from the old rate's speech and is based on the determined pitch. The number of speech samples from the old rate that are assigned to the short-term sequence is equal to the last sample minus the pitch period (P), in samples, or (N−P to N) samples of the last rate's speech. In addition, a short-term speech segment is assigned from the new rate's speech, and the number of samples is equivalent to the first frame of speech samples at the new rate.

Inblock625, the short-term speech samples from the old and new rates are correlated to determine an offset at which the short-term speech samples are most alike. A correlation matrix is formed, and the best value of likeness is found at an offset value where the correlation is at a peak. This represents the offset, or relative shift, in pitch likeness, where the old and new speech segments most likely overlap.

Inblock630, speech samples are removed from either of the short-term sequences. The area of overlap in the speech segments is removed so that redundant data is not present. For example, a portion of the old rate speech can be removed from one short-term sequence, or a portion of the new rate speech can be removed from the other short-term sequence. Inblock635, the two short term sequences are concatenated.

A vector variable is used to store the concatenated short term speech samples. If speech was removed due to the operation inblock630, then the resulting speech is to be warped (or stretched), so that the concatenated speech length in time equals the amount before any samples were removed. This process is performed so that no perceptual artifacts are audible to the human ear. A table of warping percentages exists such that the expansion factor versus pitch and length of speech can be determined. Inblock640, the expansion factor (XF) is determined based on the expansion percentage (X%) and the pitch (P) as XF=X%*P.

Inblock645, the warping factor (W_f) is set based on the expansion factor and the sample level of the buffer as shown in FIG.5. The number of samples N_DAis read from the variable buffer, and W_f, is determined as discussed above. Inblock650, the speech segment is warped; inblock655, the samples are output to the D/A; and inblock660, the D/A output is sent to an audio device or a storage device. An exemplary warping function is shown in the pseudo-code that follows:

	//	TIME_WARP warps the signal x to be stretched in time.
	//
	//	SYNTAX:  Y = TIME_WARP(X, N2, FS, TYPE);
	//	INPUTS:

	//	X = input signal to be stretched, a vector.
	//	N2 = number of sample points in desired output Y.
	//	FS = sampling frequency of the input signal Y.
	//	TYPE = type of stretching/compressing function

	//	=0 → linear
	//	=1 → bartlett stretch
	//	=2 → blackman
	//	=3 → boxcar (no stretch)
	//	=4 → hamming
	//	=5 → hanning
	//	=6 → kaiser (beta = 1)
	//	=7 → tiang
	//

	//	X is computed at a constant interval T = 1/fs
	//	compute window where, dw(n) = f(n) = window(n)
		(i.e., [dw(1), dw(2),

		. . . dw(N)] = [f(1), f(2), . . . f(N)]) where N is
		the length of X.

	//	Note: sum(dw(n))*c = n2 − N,
		where we assume we are always stretching X, so

n2 > M, and c is a normalization constant.

	//	c = (n2 − N)/sum(dw(n))
	//	dw = c*dw
	//	n_new = 1 + dw(n); n = 1, 2, 3, . . . N.
	//	Note: N*T goes out to end of X in time. n2*T goes

		out to the end of Y in time, and so
		(n2 − N)*T is the
		amount X is stretched to get Y.

	//	Find new indexes of warped X
	//	for I = 1:N

//

if I == 1;

//

ns(I) = n_new(I);

//

else

//

ns(I) = ns(I − 1) + n_new(I)

//

end

	//	end

FIG. 7 is a graph of speech distortion waveforms. Speech spectral distortion (SD), a performance metric of speech, is measured at the receiver. The SD of the adaptive-rate system (ARS)730 and the SD of a fixed-rate system (FRS)720 are shown. In this example, the FRS operates at a symbol rate of 19.2 kilosymbols/sec (ks/s), a MRSTC vocoder rate of 9.6 kb/s, with the same channel coding as utilized in the adaptive-rate system. The received signal-to-noise ratio (SNR)710 versus time characteristic of the channel is shown in FIG. 7 for a fixed-rate transmitter. Note that for the adaptive-rate system, the received SNR can be written as E_s/N_o=(C/N_o)(1/R_s), where C is the average received power, N_ois the noise spectral density, R_sis the modem symbol rate, and the transmitter power is fixed.

For both systems, the short-time SD, SD_st, is averaged over a 3-to-5 frame window of a 30 second speech sequence. The adaptive-rate system is implemented utilizing rate ½ channel coding, modem rates of 19.2/9.6/4.8/2.4 ks/s, and MRSTC rates of 9.6/4.8/2.4/1.2 kb/s. FIG. 7 shows the ARS speech quality to be superior to that of the FRS. Informal listening tests confirmed a large improvement in ARS speech quality compared to the FRS. The results in FIG. 7 show that the increase in SD at low values of E_s/N_ois due in part to a large number of bit errors entering the vocoder, and it is much greater than the increase in SD due to lower source encoding rates in the MRSTC. An increase in C/N_osystem operating range, of greater than 9 dB, can then be achieved with the ARS, given that the FRS degrades rapidly below 0 dB E_s/N_o.SD for both systems, averaged over the 30-second sequence, was 9.5 dB and 0.6 dB, respectively.

FIG. 8 is a graph of delay probabilities. An important consideration of a multi-rate vocoder is the algorithmic delay incurred when switching through a wide range of bit rates. To demonstrate the effectiveness of the VSRB method compared to a FSRB approach, a simulation has been performed modeling the delay probabilities, P(t_df) and P(t_dv). The simulation tested the switching algorithm over 50 k independent switch requests. For FSRB, B_tis fixed, with a size which is limited to an integer multiple of the vocoder frame size in bits. This means that no data is available at the output until the integer number is reached. For example, if B_tis 400 bits and B_vis 100 bits, then it does not output data until 4 frames have been stored in the SCBB. FIG. 8 shows the delay probability simulation results. The VSRB system has substantially less delay than the FSRB system.

In summary, the method and apparatus of the present invention provide a seamless rate transition mechanism in a multi-rate speech system. While we have shown and described specific embodiments of the present invention, further modifications and improvements will occur to those skilled in the art. We desire it to be understood, therefore, that this invention is not limited to the particular forms shown, and we intend in the appended claims to cover all modifications that do not depart from the spirit and scope of this invention.

Claims (19)

What is claimed is:

1. A method of coupling a first set of speech data with a second set of speech data comprising:

removing a portion of the second set of speech data to create a second subset of speech data;

concatenating the first set of speech data with the second subset of speech data to produce a concatenated set of data;

warping the concatenated set of data to create a warped concatenated set of data; and

sending the warped concatenated set of data to a digital-to-analog (D/A) converter buffer having a number of samples included therein.

2. The method ofclaim 1 wherein:

the first set of speech data has a first length in time;

the second set of speech data has a second length in time; and

the warping creates the warped concatenated set of data having a third length in time substantially equal to the sum of the first length in time and the second length in time.

3. The method ofclaim 1 wherein the first set of speech data comprises a non-integer number of first decoded frames having been previously coded at a first rate, and the second set of speech data comprises a single second decoded frame having previously been coded at a second rate.

4. The method ofclaim 3 wherein the first set of speech data represents speech having a pitch, the method further comprising:

determining the pitch of the speech represented by the first set of speech data, wherein the pitch has a period associated therewith; and

setting a size of the first set of speech data substantially equal to the period of the pitch.

5. The method ofclaim 1 wherein removing a portion of the second set of speech data comprises:

correlating the first set of speech data with the second set of speech data to determine an offset;

determining a size of the portion of the second set of speech data as a function of the offset; and

removing the portion of the second set of speech data to create the second subset of speech data.

6. The method ofclaim 1 wherein the warping is a function of the number of samples included in the D/A buffer.

7. A method of combining two speech waveforms, the method comprising:

correlating the two speech waveforms to produce an offset;

reducing a size of one of the two speech waveforms by a number of samples substantially equal to the offset;

concatenating the two speech waveforms; and

wherein the two speech waveforms comprise a first speech waveform decoded from at least one frame having been previously coded at a first rate.

8. The method ofclaim 7 wherein the two speech waveforms further comprises a second speech waveform decoded from at least one frame having been previously coded at a second rate different from the first rate.

9. The method ofclaim 8 further comprising:

prior to correlating, determining a period of a pitch of the first speech waveform;

prior to correlating, truncating the first speech waveform such that a size of the first speech waveform is substantially equal to the period of the pitch of the first speech waveform; and

prior to correlating, truncating the second speech waveform such that a size of the second speech waveform is substantially equal to a size of one of the at least one frame having been previously coded at a second rate.

10. The method ofclaim 9 wherein reducing a size of one of the two speech waveforms comprises reducing the size of the first speech waveform.

11. The method ofclaim 9 wherein reducing a size of one of the two speech waveforms comprises reducing the size of the second speech waveform.

12. The method ofclaim 7 further comprising stretching the two speech waveforms to compensate for the size of one of the two speech waveforms being reduced.

13. In a speech encoding system that encodes speech in a plurality of frames including a first frame and a second frame, each of the plurality of frames having a coding rate assigned thereto, a method of adaptively changing from a first coding rate to a second coding rate, the method comprising:

receiving a rate change request during encoding of the first frame at a first coding rate;

finishing encoding the first frame at the first coding rate;

encoding at least a portion of the first frame at the second coding rate; and

encoding the second frame at a second coding rate.

14. The method ofclaim 13 further comprising:

storing a plurality of speech samples in a speech buffer; and

marking a location within the speech buffer denoting an end of the first frame.

15. The method ofclaim 13 further comprising:

storing a plurality of speech samples in a speech buffer; and

marking a location within the speech buffer denoting a beginning of the second frame.

16. A transmitter that includes an adaptive frame rate buffer, the adaptive frame rate buffer comprising:

a source coder bit buffer configured to receive a plurality of frames of coded speech from a multi-rate source coder;

an adaptive transmit frame buffer configured to receive an integer or non-integer number of the plurality of frames from the source coder bit buffer;

the multi-rate source coder configured to code the plurality of frames of coded speech;

a speech buffer coupled to the multi-rate source coder, the speech buffer being configured to hold past samples of speech data; and

wherein the multi-rate coder is further configured to utilize the past samples of speech data when a rate change request is received.

17. A receiver comprising:

a seamless rate transition module having an input node upon which frames of decoded speech are received; and

a variable buffer having an input coupled to an output of the seamless rate transition module, the variable buffer having a number of speech samples included therein;

wherein the seamless rate transition module is configured to deposit a variable number of speech samples in the variable buffer, the variable number of speech samples being a function of the number of speech samples in the variable buffer.

18. The receiver ofclaim 17 further comprising a multi-rate source decoder having an output node coupled to the input node of the seamless rate transition module.

19. The receiver ofclaim 18 further comprising a variable size rate buffer having an output node coupled to an input node of the multi-rate source decoder.

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